[Freeswitch-users] Call Recovery when using TCP
Anthony Minessale
anthony.minessale at gmail.com
Thu Jul 25 01:13:36 MSD 2013
I think I would need to see the whole sip trace of the original invite and
the recover leg.
The only diff I see between mine and yours is mine is using FS on both
sides and its including transport=tcp
On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
> maybe its because transport=tcp is not there in the req invite, I'll see
> if I can force that into the url when applicable.
> The via from the 200ok is the one that is preserved for the recover invite.
> This is all a bit of trickery cos the sofia stack has no idea its doing a
> recovery it thinks its a new outbound uac call.
> We someday need to change the code to allow us to create a uas instance on
> demand from the dialog data we have.
>
>
>
> On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>
>> So this would make more sense. was scratching my head earlier… didn't
>> make sense
>>
>> Here's what fs gets
>>
>> INVITE sip:+353877857933 at voice.plannet21.ie:5060 SIP/2.0
>> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK42DC2424
>>
>> Here's whats sent from recovering fs
>>
>> INVITE sip:+35319032109 at 198.19.255.1:5060 SIP/2.0
>> Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKXggpr14eHNNKH
>> Route: <sip:+35319032109 at 198.19.255.1:35556;transport=tcp>
>>
>> Looking for differences the only one I can see is rport. I see on
>> fisheye, for the update, I can see reference to sip_network_port, is it
>> related, I have no experience with C. Does the initial invite need to have
>> rport in the via?
>> As a test I added rport but still the same behaviour, although 33333 is
>> not real and I just statically set it outgoing from uac, but this causes
>> subsequent packets to update and have the rport present
>>
>> FS in
>>
>> INVITE sip:+353877857933 at voice.plannet21.ie:5060 SIP/2.0
>> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=33333
>>
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=13129
>>
>> FS Recover
>>
>> INVITE sip:+35319032109 at 198.19.255.1:5060 SIP/2.0
>> Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKKB0t2By0am6HB
>> Route: <sip:+35319032109 at 198.19.255.1:13129;transport=tcp>
>>
>>
>> On 24 Jul 2013, at 18:18, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>
>> Interesting. In the test I did when making the patch, the recover INVITE
>> was tcp, it depends heavily on the VIA header on the original invite having
>> TCP present in it.
>>
>> send 1281 bytes to tcp/[1.x.x.x]:5060 at 17:14:42.035251:
>>
>> ------------------------------------------------------------------------
>> INVITE sip:mod_sofia at 1.x.x.x:5060 SIP/2.0
>> Via: SIP/2.0/TCP 1.x.x.x;branch=z9hG4bKS0Q3SKXvB2Q9r;rport=50938
>> Route: <sip:1004 at 1.x.x.x:50938;transport=tcp>
>>
>>
>>
>> On Wed, Jul 24, 2013 at 5:34 AM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>
>>> Thanks Anthony, yes latest head worked.
>>>
>>> A leg still sends the recovery INVITE as UDP however now it has the
>>> route header with no loose routing.
>>>
>>> Route: <sip:+35314611947 at 198.19.255.1:61767;transport=tcp>
>>>
>>> So I'm assuming the UAC uses this info to find session in UAC Table and
>>> reestablishes the session, but as UDP, even though UAC specifically is told
>>> to only use TCP.
>>>
>>> To test I put a firewall in the path and blocked UDP 5060 and the call
>>> failed to recover, as expected.
>>>
>>> So I think for this to work UDP will still need to be open on the UAC to
>>> recover the call.
>>>
>>> As you said it won't work for all UACs or all situations but its a
>>> workable solution.
>>>
>>> For reference this works when UAC is either Cisco or Dialogic.
>>>
>>>
>>> On 23 Jul 2013, at 19:53, Anthony Minessale <anthony.minessale at gmail.com>
>>> wrote:
>>>
>>> Try latest head, no promises on every endpoint.
>>> P.S. try Jira next time.
>>>
>>>
>>>
>>> On Tue, Jul 23, 2013 at 11:24 AM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>
>>>> I have pasted up the logs from a test call A leg UDP, B leg TCP that
>>>> recovered ok
>>>>
>>>> initial call - fs crashed
>>>> http://pastebin.com/0xe0QyFC
>>>>
>>>> recovered call
>>>> http://pastebin.com/ByjJ4nhf
>>>>
>>>>
>>>> On 23 Jul 2013, at 16:39, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> Is the B-leg the same call, or a new call?
>>>>
>>>>
>>>>
>>>> On 23 July 2013 16:31, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>>>
>>>>> Thanks Brian,
>>>>>
>>>>> Initially I though the same and looked for something to migrate the
>>>>> tcp session, tcpcp and sockmi, but no joy.
>>>>>
>>>>> I was doing some more testing and noticed that if the B leg was TCP
>>>>> and the A leg UDP the call recovered.
>>>>>
>>>>> So I though my earlier assumption about TCP connection dropping was
>>>>> wrong as it seems the B leg reestablishes the session on the recovering
>>>>> server.
>>>>> The issue just seems to be the recovery of the A leg. FS always sends
>>>>> the A leg recovery INVITE as UDP. Even if original call was TCP. If I could
>>>>> force it to use TCP I believe it would recover the call.
>>>>>
>>>>> Below is a call with A leg as UDP and B leg as TCP thats recovers fine.
>>>>>
>>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1030 New Channel
>>>>> sofia/private/+35319032109 at 198
>>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1028 Rename
>>>>> Channel sofia/private/+35319032109@
>>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_core_sqldb.c:2744
>>>>> Resurrecting fallen channel sofia/priva
>>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1030 New Channel
>>>>> sofia/internal/0877857933 at 10.1
>>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename
>>>>> Channel sofia/internal/0877857933 at 1
>>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_core_sqldb.c:2744
>>>>> Resurrecting fallen channel sofia/inter
>>>>> send 1110 bytes to udp/[10.101.23.203]:5060 at 15:19:46.459562:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> INVITE sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS
>>>>> Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>>>>> Max-Forwards: 70
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> CSeq: 46960329 INVITE
>>>>> Contact: <sip:0877857933 at 10.101.23.110:5060>
>>>>> User-Agent: LAB - SBC
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY
>>>>> Supported: timer, precondition, path, replaces
>>>>> Allow-Events: talk, hold, conference, refer
>>>>> Privacy: none
>>>>> Content-Type: application/sdp
>>>>> Content-Disposition: session
>>>>> Content-Length: 246
>>>>> X-FS-Support: update_display,send_info
>>>>> P-Asserted-Identity: "ae019032109" <sip:ae019032109 at 10.101.23.203>
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1374564092 1374564094 IN IP4 10.101.24.110
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 10.101.24.110
>>>>> t=0 0
>>>>> m=audio 28694 RTP/AVP 8 101 13
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=rtpmap:13 CN/8000
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 398 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463116:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 100 Trying
>>>>> Via: SIP/2.0/UDP
>>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> CSeq: 46960329 INVITE
>>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463966:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>>> CSeq: 46960329 INVITE
>>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>>> NOTIFY, INFO, REFER, UPDATE
>>>>> Supported: path, replaces, timer, tdialog
>>>>> Require: timer
>>>>> Session-Expires: 1800;refresher=uas
>>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 239
>>>>>
>>>>> v=0
>>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>>> s=Dialogic-SIP
>>>>> c=IN IP4 10.101.24.203
>>>>> t=0 0
>>>>> m=audio 8332 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-15
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:47.101507:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>>> CSeq: 46960329 INVITE
>>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>>> NOTIFY, INFO, REFER, UPDATE
>>>>> Supported: path, replaces, timer, tdialog
>>>>> Require: timer
>>>>> Session-Expires: 1800;refresher=uas
>>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 239
>>>>>
>>>>> v=0
>>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>>> s=Dialogic-SIP
>>>>> c=IN IP4 10.101.24.203
>>>>> t=0 0
>>>>> m=audio 8332 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-15
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:48.070746:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>>> CSeq: 46960329 INVITE
>>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>>> NOTIFY, INFO, REFER, UPDATE
>>>>> Supported: path, replaces, timer, tdialog
>>>>> Require: timer
>>>>> Session-Expires: 1800;refresher=uas
>>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 239
>>>>>
>>>>> v=0
>>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>>> s=Dialogic-SIP
>>>>> c=IN IP4 10.101.24.203
>>>>> t=0 0
>>>>> m=audio 8332 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-15
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 1175 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.473346:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> INVITE sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q
>>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKZ21tSFUQvy38c
>>>>> Max-Forwards: 69
>>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>>> >;tag=BK8QvjBeDHXtN
>>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>>> CSeq: 46960329 INVITE
>>>>> Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>>>>> User-Agent: PlanNet21 Communications - SBC
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY
>>>>> Supported: timer, precondition, path, replaces
>>>>> Allow-Events: talk, hold, conference, refer
>>>>> Privacy: none
>>>>> Content-Type: application/sdp
>>>>> Content-Disposition: session
>>>>> Content-Length: 246
>>>>> X-FS-Support: update_display,send_info
>>>>> P-Asserted-Identity: "+353877857933" <
>>>>> sip:+353877857933 at 78.158.110.24>
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1374569678 1374569680 IN IP4 78.158.110.24
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 78.158.110.24
>>>>> t=0 0
>>>>> m=audio 23108 RTP/AVP 8 101 13
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=rtpmap:13 CN/8000
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 484 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.536636:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 100 Trying
>>>>> Via: SIP/2.0/TCP
>>>>> 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP
>>>>> 78.158.110.24;rport;
>>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>>> >;tag=BK8QvjBeDHXtN
>>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>>> Date: Tue, 23 Jul 2013 15:19:49 GMT
>>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>>> CSeq: 46960329 INVITE
>>>>> Allow-Events: telephone-event
>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> recv 1073 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.601460:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/TCP
>>>>> 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP
>>>>> 78.158.110.24;rport;
>>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>>> >;tag=BK8QvjBeDHXtN
>>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>>> Date: Tue, 23 Jul 2013 15:19:49 GMT
>>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>>> CSeq: 46960329 INVITE
>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>> Allow-Events: telephone-event
>>>>> Remote-Party-ID: "Ian McGrath" <sip:+1947 at 198.19.255.1
>>>>> >;party=called;screen=yes;privacy=off
>>>>> Contact: <sip:+35319032109 at 198.19.255.1:5060;transport=tcp>
>>>>> Supported: replaces
>>>>> Supported: sdp-anat
>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>> Supported: timer
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 247
>>>>>
>>>>> v=0
>>>>> o=CiscoSystemsSIP-GW-UserAgent 3902 8614 IN IP4 198.19.255.1
>>>>> s=SIP Call
>>>>> c=IN IP4 198.19.255.1
>>>>> t=0 0
>>>>> m=audio 19050 RTP/AVP 8 101
>>>>> c=IN IP4 198.19.255.1
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 440 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.603288:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> ACK sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKcy8r8S6SraZtK
>>>>> Max-Forwards: 70
>>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>>> >;tag=BK8QvjBeDHXtN
>>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>>> CSeq: 46960329 ACK
>>>>> Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> send 466 bytes to udp/[10.101.23.203]:5060 at 15:19:49.606747:
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> ACK sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m
>>>>> Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>>>>> Max-Forwards: 70
>>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>>> To: <sip:0877857933 at 10.101.23.203
>>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>>> CSeq: 46960329 ACK
>>>>> Contact: <sip:0877857933 at 10.101.23.110:5060>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 23 Jul 2013, at 14:47, Brian West <brian at freeswitch.org> wrote:
>>>>>
>>>>> > You can't do call recovery on TCP at the moment, You have no way to
>>>>> re-establish the TCP connections once FreeSWITCH goes down.
>>>>> >
>>>>> > /b
>>>>> >
>>>>> > Em Jul 23, 2013, às 8:01 AM, Anthony McGarry <agtmcgarry at gmail.com>
>>>>> escreveu:
>>>>> >
>>>>> >> Anyone using TCP in this scenario? Cannot find what I'm missing.
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
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>>>>> >
>>>>> >
>>>>> >
>>>>> >
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>>>>>
>>>>>
>>>>> _________________________________________________________________________
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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>>
>>>>>
>>>>>
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>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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